Whisker, Peter
2004-Oct-28 07:28 UTC
[Asterisk-Users] Re: call progress - what are the sticking po ints?
It looks for tones (currently hardwired as US). I have updated to include UK tones but is hard to get it to reliably recognise. For example the tones in the switch here at work are 5-10% off frequency. Correcting for this, and doing a lot of fiddling it did recognise the tones but was unreliable. I have a problem in that our office switch clears to dialtone rather than busy if the other end hangs up. I would like a way of recognising unexpected dialtone and hanging-up. So far, this has not been easy. I have changed the busydetect to clear if it gets continupus tone for 8 seconds but this does false hangups and would be useless for a fax machine. Peter -----Original Message----- From: Steve Underwood [mailto:steveu@coppice.org] Sent: 28 October 2004 14:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: call progress - what are the sticking points? Stephen David wrote:>i don't have a specific bug in mind, i was just wondering WHY call progressdoesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :)> >Back in the days of Stowger exchanges you knew when the called party answered, by a reversal of the DC voltage on your analogue line. With digital exchanges that stopped, and no solid feedback is given to the caller on ordinary analogue lines. You have to infer that someone has answered, and the reliability of that can be poor. Digital lines, like ISDNand SS7, and protocols like MFC/R2 tell you positively that someone has answered.>>I have the same problem. >>callprogress is very experimental and buggy now. >>and i've lost the .call files feature of asterisk. >>what do you think about submitting a bug on bugs.digium.com? >> >> >> > >not sure what you mean by 'lost the .call files feature', but if you have aspecific bug to post, i think it would be great if you posted it.> > > >> regards, >> shabanip >> >> > Hello, >> > >> > I've been experimenting with the call progress analysis features of *, >> > with mixed success on Zap as well as IAX channels. I've read all the >> > posts about it, including (but not limited to) >> > http://www.voip-info.org/wiki-Asterisk+auto-dial+out and the pages it >> > references. >> > >> > My question is, what's the current state -- is there any work inprogress>> > right now to improve the reliability of * call progress detection?last I>> > saw it was still listed as 'experimental'. >> > >> > What are the "problems" that are preventing a more robustimplementation>> > of call progress detection? Would this work better with different >> > hardware (ie. I've had success in the past using Dialogic telephony >> > boards)? Or is this primarily a software issue with *? >> >>If you had good results with Dialogic it was merely luck. Because they have to infer the phone has been answered, their detection only works if the calls follow their model of how someone answers the phone. Depending on your circumstances, and the nature of the calls you make, it can be hopelessly unreliable. Steve _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you.